Add support for Opus audio format

1 files changed, 229 insertions, 0 deletions

diff --git a/drivers/opus/silk/stereo_LR_to_MS.c b/drivers/opus/silk/stereo_LR_to_MS.c
new file mode 100644
index 0000000000..678f46984b
--- /dev/null
+++ b/drivers/opus/silk/stereo_LR_to_MS.c
@@ -0,0 +1,229 @@
+/***********************************************************************
+Copyright (c) 2006-2011, Skype Limited. All rights reserved.
+Redistribution and use in source and binary forms, with or without
+modification, are permitted provided that the following conditions
+are met:
+- Redistributions of source code must retain the above copyright notice,
+this list of conditions and the following disclaimer.
+- Redistributions in binary form must reproduce the above copyright
+notice, this list of conditions and the following disclaimer in the
+documentation and/or other materials provided with the distribution.
+- Neither the name of Internet Society, IETF or IETF Trust, nor the
+names of specific contributors, may be used to endorse or promote
+products derived from this software without specific prior written
+permission.
+THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS"
+AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
+IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
+ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE
+LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR
+CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF
+SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS
+INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN
+CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE)
+ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE
+POSSIBILITY OF SUCH DAMAGE.
+***********************************************************************/
+
+#ifdef OPUS_HAVE_CONFIG_H
+#include "opus_config.h"
+#endif
+
+#include "silk_main.h"
+#include "stack_alloc.h"
+
+/* Convert Left/Right stereo signal to adaptive Mid/Side representation */
+void silk_stereo_LR_to_MS(
+    stereo_enc_state            *state,                         /* I/O  State                                       */
+    opus_int16                  x1[],                           /* I/O  Left input signal, becomes mid signal       */
+    opus_int16                  x2[],                           /* I/O  Right input signal, becomes side signal     */
+    opus_int8                   ix[ 2 ][ 3 ],                   /* O    Quantization indices                        */
+    opus_int8                   *mid_only_flag,                 /* O    Flag: only mid signal coded                 */
+    opus_int32                  mid_side_rates_bps[],           /* O    Bitrates for mid and side signals           */
+    opus_int32                  total_rate_bps,                 /* I    Total bitrate                               */
+    opus_int                    prev_speech_act_Q8,             /* I    Speech activity level in previous frame     */
+    opus_int                    toMono,                         /* I    Last frame before a stereo->mono transition */
+    opus_int                    fs_kHz,                         /* I    Sample rate (kHz)                           */
+    opus_int                    frame_length                    /* I    Number of samples                           */
+)
+{
+    opus_int   n, is10msFrame, denom_Q16, delta0_Q13, delta1_Q13;
+    opus_int32 sum, diff, smooth_coef_Q16, pred_Q13[ 2 ], pred0_Q13, pred1_Q13;
+    opus_int32 LP_ratio_Q14, HP_ratio_Q14, frac_Q16, frac_3_Q16, min_mid_rate_bps, width_Q14, w_Q24, deltaw_Q24;
+    VARDECL( opus_int16, side );
+    VARDECL( opus_int16, LP_mid );
+    VARDECL( opus_int16, HP_mid );
+    VARDECL( opus_int16, LP_side );
+    VARDECL( opus_int16, HP_side );
+    opus_int16 *mid = &x1[ -2 ];
+    SAVE_STACK;
+
+    ALLOC( side, frame_length + 2, opus_int16 );
+    /* Convert to basic mid/side signals */
+    for( n = 0; n < frame_length + 2; n++ ) {
+        sum  = x1[ n - 2 ] + (opus_int32)x2[ n - 2 ];
+        diff = x1[ n - 2 ] - (opus_int32)x2[ n - 2 ];
+        mid[  n ] = (opus_int16)silk_RSHIFT_ROUND( sum, 1 );
+        side[ n ] = (opus_int16)silk_SAT16( silk_RSHIFT_ROUND( diff, 1 ) );
+    }
+
+    /* Buffering */
+    silk_memcpy( mid,  state->sMid,  2 * sizeof( opus_int16 ) );
+    silk_memcpy( side, state->sSide, 2 * sizeof( opus_int16 ) );
+    silk_memcpy( state->sMid,  &mid[  frame_length ], 2 * sizeof( opus_int16 ) );
+    silk_memcpy( state->sSide, &side[ frame_length ], 2 * sizeof( opus_int16 ) );
+
+    /* LP and HP filter mid signal */
+    ALLOC( LP_mid, frame_length, opus_int16 );
+    ALLOC( HP_mid, frame_length, opus_int16 );
+    for( n = 0; n < frame_length; n++ ) {
+        sum = silk_RSHIFT_ROUND( silk_ADD_LSHIFT( mid[ n ] + mid[ n + 2 ], mid[ n + 1 ], 1 ), 2 );
+        LP_mid[ n ] = sum;
+        HP_mid[ n ] = mid[ n + 1 ] - sum;
+    }
+
+    /* LP and HP filter side signal */
+    ALLOC( LP_side, frame_length, opus_int16 );
+    ALLOC( HP_side, frame_length, opus_int16 );
+    for( n = 0; n < frame_length; n++ ) {
+        sum = silk_RSHIFT_ROUND( silk_ADD_LSHIFT( side[ n ] + side[ n + 2 ], side[ n + 1 ], 1 ), 2 );
+        LP_side[ n ] = sum;
+        HP_side[ n ] = side[ n + 1 ] - sum;
+    }
+
+    /* Find energies and predictors */
+    is10msFrame = frame_length == 10 * fs_kHz;
+    smooth_coef_Q16 = is10msFrame ?
+        SILK_FIX_CONST( STEREO_RATIO_SMOOTH_COEF / 2, 16 ) :
+        SILK_FIX_CONST( STEREO_RATIO_SMOOTH_COEF,     16 );
+    smooth_coef_Q16 = silk_SMULWB( silk_SMULBB( prev_speech_act_Q8, prev_speech_act_Q8 ), smooth_coef_Q16 );
+
+    pred_Q13[ 0 ] = silk_stereo_find_predictor( &LP_ratio_Q14, LP_mid, LP_side, &state->mid_side_amp_Q0[ 0 ], frame_length, smooth_coef_Q16 );
+    pred_Q13[ 1 ] = silk_stereo_find_predictor( &HP_ratio_Q14, HP_mid, HP_side, &state->mid_side_amp_Q0[ 2 ], frame_length, smooth_coef_Q16 );
+    /* Ratio of the norms of residual and mid signals */
+    frac_Q16 = silk_SMLABB( HP_ratio_Q14, LP_ratio_Q14, 3 );
+    frac_Q16 = silk_min( frac_Q16, SILK_FIX_CONST( 1, 16 ) );
+
+    /* Determine bitrate distribution between mid and side, and possibly reduce stereo width */
+    total_rate_bps -= is10msFrame ? 1200 : 600;      /* Subtract approximate bitrate for coding stereo parameters */
+    if( total_rate_bps < 1 ) {
+        total_rate_bps = 1;
+    }
+    min_mid_rate_bps = silk_SMLABB( 2000, fs_kHz, 900 );
+    silk_assert( min_mid_rate_bps < 32767 );
+    /* Default bitrate distribution: 8 parts for Mid and (5+3*frac) parts for Side. so: mid_rate = ( 8 / ( 13 + 3 * frac ) ) * total_ rate */
+    frac_3_Q16 = silk_MUL( 3, frac_Q16 );
+    mid_side_rates_bps[ 0 ] = silk_DIV32_varQ( total_rate_bps, SILK_FIX_CONST( 8 + 5, 16 ) + frac_3_Q16, 16+3 );
+    /* If Mid bitrate below minimum, reduce stereo width */
+    if( mid_side_rates_bps[ 0 ] < min_mid_rate_bps ) {
+        mid_side_rates_bps[ 0 ] = min_mid_rate_bps;
+        mid_side_rates_bps[ 1 ] = total_rate_bps - mid_side_rates_bps[ 0 ];
+        /* width = 4 * ( 2 * side_rate - min_rate ) / ( ( 1 + 3 * frac ) * min_rate ) */
+        width_Q14 = silk_DIV32_varQ( silk_LSHIFT( mid_side_rates_bps[ 1 ], 1 ) - min_mid_rate_bps,
+            silk_SMULWB( SILK_FIX_CONST( 1, 16 ) + frac_3_Q16, min_mid_rate_bps ), 14+2 );
+        width_Q14 = silk_LIMIT( width_Q14, 0, SILK_FIX_CONST( 1, 14 ) );
+    } else {
+        mid_side_rates_bps[ 1 ] = total_rate_bps - mid_side_rates_bps[ 0 ];
+        width_Q14 = SILK_FIX_CONST( 1, 14 );
+    }
+
+    /* Smoother */
+    state->smth_width_Q14 = (opus_int16)silk_SMLAWB( state->smth_width_Q14, width_Q14 - state->smth_width_Q14, smooth_coef_Q16 );
+
+    /* At very low bitrates or for inputs that are nearly amplitude panned, switch to panned-mono coding */
+    *mid_only_flag = 0;
+    if( toMono ) {
+        /* Last frame before stereo->mono transition; collapse stereo width */
+        width_Q14 = 0;
+        pred_Q13[ 0 ] = 0;
+        pred_Q13[ 1 ] = 0;
+        silk_stereo_quant_pred( pred_Q13, ix );
+    } else if( state->width_prev_Q14 == 0 &&
+        ( 8 * total_rate_bps < 13 * min_mid_rate_bps || silk_SMULWB( frac_Q16, state->smth_width_Q14 ) < SILK_FIX_CONST( 0.05, 14 ) ) )
+    {
+        /* Code as panned-mono; previous frame already had zero width */
+        /* Scale down and quantize predictors */
+        pred_Q13[ 0 ] = silk_RSHIFT( silk_SMULBB( state->smth_width_Q14, pred_Q13[ 0 ] ), 14 );
+        pred_Q13[ 1 ] = silk_RSHIFT( silk_SMULBB( state->smth_width_Q14, pred_Q13[ 1 ] ), 14 );
+        silk_stereo_quant_pred( pred_Q13, ix );
+        /* Collapse stereo width */
+        width_Q14 = 0;
+        pred_Q13[ 0 ] = 0;
+        pred_Q13[ 1 ] = 0;
+        mid_side_rates_bps[ 0 ] = total_rate_bps;
+        mid_side_rates_bps[ 1 ] = 0;
+        *mid_only_flag = 1;
+    } else if( state->width_prev_Q14 != 0 &&
+        ( 8 * total_rate_bps < 11 * min_mid_rate_bps || silk_SMULWB( frac_Q16, state->smth_width_Q14 ) < SILK_FIX_CONST( 0.02, 14 ) ) )
+    {
+        /* Transition to zero-width stereo */
+        /* Scale down and quantize predictors */
+        pred_Q13[ 0 ] = silk_RSHIFT( silk_SMULBB( state->smth_width_Q14, pred_Q13[ 0 ] ), 14 );
+        pred_Q13[ 1 ] = silk_RSHIFT( silk_SMULBB( state->smth_width_Q14, pred_Q13[ 1 ] ), 14 );
+        silk_stereo_quant_pred( pred_Q13, ix );
+        /* Collapse stereo width */
+        width_Q14 = 0;
+        pred_Q13[ 0 ] = 0;
+        pred_Q13[ 1 ] = 0;
+    } else if( state->smth_width_Q14 > SILK_FIX_CONST( 0.95, 14 ) ) {
+        /* Full-width stereo coding */
+        silk_stereo_quant_pred( pred_Q13, ix );
+        width_Q14 = SILK_FIX_CONST( 1, 14 );
+    } else {
+        /* Reduced-width stereo coding; scale down and quantize predictors */
+        pred_Q13[ 0 ] = silk_RSHIFT( silk_SMULBB( state->smth_width_Q14, pred_Q13[ 0 ] ), 14 );
+        pred_Q13[ 1 ] = silk_RSHIFT( silk_SMULBB( state->smth_width_Q14, pred_Q13[ 1 ] ), 14 );
+        silk_stereo_quant_pred( pred_Q13, ix );
+        width_Q14 = state->smth_width_Q14;
+    }
+
+    /* Make sure to keep on encoding until the tapered output has been transmitted */
+    if( *mid_only_flag == 1 ) {
+        state->silent_side_len += frame_length - STEREO_INTERP_LEN_MS * fs_kHz;
+        if( state->silent_side_len < LA_SHAPE_MS * fs_kHz ) {
+            *mid_only_flag = 0;
+        } else {
+            /* Limit to avoid wrapping around */
+            state->silent_side_len = 10000;
+        }
+    } else {
+        state->silent_side_len = 0;
+    }
+
+    if( *mid_only_flag == 0 && mid_side_rates_bps[ 1 ] < 1 ) {
+        mid_side_rates_bps[ 1 ] = 1;
+        mid_side_rates_bps[ 0 ] = silk_max_int( 1, total_rate_bps - mid_side_rates_bps[ 1 ]);
+    }
+
+    /* Interpolate predictors and subtract prediction from side channel */
+    pred0_Q13  = -state->pred_prev_Q13[ 0 ];
+    pred1_Q13  = -state->pred_prev_Q13[ 1 ];
+    w_Q24      =  silk_LSHIFT( state->width_prev_Q14, 10 );
+    denom_Q16  = silk_DIV32_16( (opus_int32)1 << 16, STEREO_INTERP_LEN_MS * fs_kHz );
+    delta0_Q13 = -silk_RSHIFT_ROUND( silk_SMULBB( pred_Q13[ 0 ] - state->pred_prev_Q13[ 0 ], denom_Q16 ), 16 );
+    delta1_Q13 = -silk_RSHIFT_ROUND( silk_SMULBB( pred_Q13[ 1 ] - state->pred_prev_Q13[ 1 ], denom_Q16 ), 16 );
+    deltaw_Q24 =  silk_LSHIFT( silk_SMULWB( width_Q14 - state->width_prev_Q14, denom_Q16 ), 10 );
+    for( n = 0; n < STEREO_INTERP_LEN_MS * fs_kHz; n++ ) {
+        pred0_Q13 += delta0_Q13;
+        pred1_Q13 += delta1_Q13;
+        w_Q24   += deltaw_Q24;
+        sum = silk_LSHIFT( silk_ADD_LSHIFT( mid[ n ] + mid[ n + 2 ], mid[ n + 1 ], 1 ), 9 );    /* Q11 */
+        sum = silk_SMLAWB( silk_SMULWB( w_Q24, side[ n + 1 ] ), sum, pred0_Q13 );               /* Q8  */
+        sum = silk_SMLAWB( sum, silk_LSHIFT( (opus_int32)mid[ n + 1 ], 11 ), pred1_Q13 );       /* Q8  */
+        x2[ n - 1 ] = (opus_int16)silk_SAT16( silk_RSHIFT_ROUND( sum, 8 ) );
+    }
+
+    pred0_Q13 = -pred_Q13[ 0 ];
+    pred1_Q13 = -pred_Q13[ 1 ];
+    w_Q24     =  silk_LSHIFT( width_Q14, 10 );
+    for( n = STEREO_INTERP_LEN_MS * fs_kHz; n < frame_length; n++ ) {
+        sum = silk_LSHIFT( silk_ADD_LSHIFT( mid[ n ] + mid[ n + 2 ], mid[ n + 1 ], 1 ), 9 );    /* Q11 */
+        sum = silk_SMLAWB( silk_SMULWB( w_Q24, side[ n + 1 ] ), sum, pred0_Q13 );               /* Q8  */
+        sum = silk_SMLAWB( sum, silk_LSHIFT( (opus_int32)mid[ n + 1 ], 11 ), pred1_Q13 );       /* Q8  */
+        x2[ n - 1 ] = (opus_int16)silk_SAT16( silk_RSHIFT_ROUND( sum, 8 ) );
+    }
+    state->pred_prev_Q13[ 0 ] = (opus_int16)pred_Q13[ 0 ];
+    state->pred_prev_Q13[ 1 ] = (opus_int16)pred_Q13[ 1 ];
+    state->width_prev_Q14     = (opus_int16)width_Q14;
+    RESTORE_STACK;
+}